[Asterisk-video] MP4Play async Audio / Video
Thomas Z.
mobilemail at gmx-topmail.de
Tue Sep 25 08:52:16 CDT 2007
Hi Sergio,
thank you very much. The result with the Helix Mobile Producer is much
better.
But we need a software runs on a Linux environment.
Can you support us with a another linux software?
Regards
Thomas
Sergio Garcia schrieb:
> You coulde try with qt encoder or the helix mobile producer..
>
> ---------- Original Message ----------------------------------
> From: "Thomas Z." <mobilemail at gmx-topmail.de>
> Reply-To: Development discussion of video media support in Asterisk<asterisk-video at lists.digium.com>
> Date: Tue, 25 Sep 2007 10:52:07 +0200
>
>
>> Hello Sergio,
>>
>> do you know a freeware or cheap CBR encoder?
>>
>> Regards
>> Thomas
>>
>> Sergio Garcia Murillo schrieb:
>>
>>> Hi Thomas, Thomas and Ratmin.. :)
>>>
>>> Back to work form a long abroad weekend (nice Belgium!)
>>>
>>> As Ratmin has stated the Skew indication would help to indicate the
>>> handset the delay between audio and video.
>>> It's usually used to correct fixed delays introduced by internal
>>> jitter buffers and thinks alike. The problem that Thomas
>>> Z. is suffering seems to be produced by using ffmpeg for video
>>> encoding (as Thomas without Z points :)
>>> The problem is the following, ffmpeg is a great VBR (variable bit
>>> rate) encoder, but I've not been able to behave like a
>>> CBR (constant bit rate) encoder. The problem is that it's prepared for
>>> movies, not for streaming, and defenitevilly not
>>> at this low bitrates.
>>> The inner loop of the encoder always tries to encode ALL macroblocks
>>> in a frame, so in I frames and at a very low
>>> bitrate it simply just can't get a small amount of data for that frame
>>> and when it's enqued into h324m it takes more
>>> than one frame time to send (sometimes even more than a second), the
>>> following frames are small and sent after
>>> that until the queue is empty, which causes a fast-forward effect
>>> on some handsets. As you can imagine adjustime
>>> audio to that scenary is jus impossible (even with the Skew Indication
>>> the result would be horrible!!)
>>>
>>> The good solution: use a CBR encoder :)
>>> The not so good solution: drop packets when the queue is filled, but
>>> you'll end up dropping all the frames between
>>> each I frame.. but in that case it would be much better to lower the
>>> video fps...
>>>
>>> BR
>>> Sergio
>>>
>>> ----- Original Message -----
>>> *From:* Ramtin Amin <mailto:keytwho at hotmail.com>
>>> *To:* Development discussion of video media support in Asterisk
>>> <mailto:asterisk-video at lists.digium.com>
>>> *Sent:* Monday, September 24, 2007 2:01 PM
>>> *Subject:* Re: [Asterisk-video] MP4Play async Audio / Video
>>>
>>>
>>>
>>>
>>> RTSP has timestamp... so you can usually know how to sync the
>>> audio/video
>>> But the problem would be more that we'll need to add a Skew
>>> Indication concept inside asterisk's core so it could make the
>>> message go throught the different channel/application, as it is
>>> currently done for VideoFastUpdatePicutre... Which means that
>>> currently, if a SIP channel receives a VideoFastUpdatePicture, it
>>> is capable of resending it to an other channel...
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> > Date: Mon, 24 Sep 2007 13:48:54 +0200
>>> > From: mobilemail at gmx-topmail.de
>>> > To: asterisk-video at lists.digium.com
>>> > Subject: Re: [Asterisk-video] MP4Play async Audio / Video
>>> >
>>> > Hello Ramin,
>>> > hello Thomas.
>>> >
>>> > I think both ways are possible. But I don't know if it easy to
>>> implement.
>>> > Sergio, what do you think about it?
>>> >
>>> > Is this also a problem if we stream the audi/video via rtsp?
>>> >
>>> > Regrads
>>> > Thomas
>>> >
>>> >
>>> > Ramtin Amin schrieb:
>>> > > hello
>>> > >
>>> > > The way I solved this problem was by sending a H223SkewIndication
>>> > > message to the other terminal
>>> > > Acutally, You will have to ask Sergio to add a H245 indication
>>> message
>>> > > of type h223SkewIndication and when playing the video with
>>> mp4play, he
>>> > > will have to see the difference of timing between Video and
>>> Audio and
>>> > > then send this value to the remote terminal so the lip sync
>>> would work...
>>> > >
>>> > >
>>> > >
>>> > > **
>>> > >
>>> > > *2.3 Multipoint Lip Synchronization*
>>> > >
>>> > > In a multipoint VC, each terminal may transmit different
>>> > >
>>> > > //
>>> > >
>>> > > /H223SkewIndication /message for associated video and
>>> > >
>>> > > audio channels in H.223 protocol. To enable lip
>>> > >
>>> > > synchronization at receiving terminals, MCUs will
>>> > >
>>> > > transmit accurate /H223SkewIndication /messages. MCUs
>>> > >
>>> > > may accomplish this by adding delay to equalize the
>>> > >
>>> > > audio/video skew for all transmitting terminals. When
>>> > >
>>> > > switching between broadcasting terminals, H.223 may
>>> > >
>>> > > transmit a new /H223SkewIndication /message reflecting the
>>> > >
>>> > > audio/video skew of the current broadcaster.
>>> > >
>>> > >
>>> > > --
>>> > > Ramtin Amin
>>> > >
>>> > >
>>> > >
>>> > >
>>> > >
>>> ------------------------------------------------------------------------
>>> > >
>>> > > > From: thomas.frieling at viif.de
>>> > > > To: asterisk-video at lists.digium.com
>>> > > > Date: Mon, 24 Sep 2007 11:48:12 +0200
>>> > > > Subject: Re: [Asterisk-video] MP4Play async Audio / Video
>>> > > >
>>> > > > Hi Thomas Z!
>>> > > >
>>> > > > I think this problem is due to the bitrate restriction on 3G
>>> calls. The
>>> > > > audio stream is always sent immediately while the video
>>> stream has to
>>> > > > use what is left of the bandwidth. This is why videos become
>>> synchrous
>>> > > > again when the video bitrate is pretty low for a while in
>>> the video...
>>> > > >
>>> > > > Take a look at this discussion:
>>> > > >
>>> > >
>>> http://lists.digium.com/pipermail/asterisk-video/2007-September/001257.html
>>> > > >
>>> > > > One solution is to reencode every video with a low bitrate
>>> and use a
>>> > > > fixed bitrate encoder (ffmpeg is dynamic bitrate for
>>> example). I had the
>>> > > > impression though that this still doesn't always work,
>>> especially when
>>> > > > the UMTS connection is not too good...
>>> > > >
>>> > > > My idea is that we check each time before sending a keyframe
>>> if there is
>>> > > > already a new keyframe in the queue. If this is the case, we
>>> jump to the
>>> > > > most recent keyframe and just drop the data before that.
>>> > > >
>>> > > > What do you think about this? How hard to implement?
>>> > > >
>>> > > > Regards,
>>> > > > Thomas F
>>> > > >
>>> > > >
>>> > > >
>>> > > > Am Montag, den 24.09.2007, 11:15 +0200 schrieb Thomas Z.:
>>> > > > > Hello,
>>> > > > >
>>> > > > > We have the problem, that a converted mp4 file is
>>> asynchronous via 3g
>>> > > > > network.
>>> > > > > If we play the file on a pc, everything is synchron.
>>> > > > > We tried already to reduce the quality and framerate of
>>> the video. But
>>> > > > > nothing helps.
>>> > > > >
>>> > > > > Is it a problem with mp4play or with the video?
>>> > > > >
>>> > > > > What can we do to get the audio and video synchronized via
>>> 3g network?
>>> > > > >
>>> > > > > Thank you,
>>> > > > > best regards
>>> > > > > Thomas
>>> > > > >
>>> > > > >
>>> > > > >
>>> > > > > _______________________________________________
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>>> > > > --
>>> > > > www.ViiF.de - your Mobile Video Community
>>> > > >
>>> > > > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>>> > > >
>>> > > > Thomas Frieling - IT Development
>>> > > > ViiF Mobile Video GmbH, Poststr. 21-22, 10178 Berlin
>>> > > > Cell: +49 (0) 173 63 62 62 3
>>> > > >
>>> > > > mailto:thomas at ViiF.de
>>> > > >
>>> > > > Sitz: Berlin, Amtgericht Berlin-Charlottenburg, HRB: 108350B
>>> > > >
>>> > > > Geschäftsführer: Daniel Höpfner, Steffen Brünn
>>> > > >
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